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Summary:ASTERISK-06401: SIP_CODEC not working for 182 Session Progress / Early media
Reporter:George Papadopoulos (gpapadop73)Labels:
Date Opened:2006-02-23 07:00:05.000-0600Date Closed:2008-01-15 17:21:46.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) fix_sip_codec_variable.patch
( 1) sipcodecearlymedia.txt
( 2) trunk-sip-codec-in-early-media.txt
Description:exten => s,1,Set(SIP_CODEC=g729)
exten => s,2,Playback(the-party-you-are-calling|noanswer)
exten => s,3,Playback(is-curntly-unavail|noanswer)

In the above case, the g729 is not used.
   -- Executing Set("SIP/213.5.1.6-40a2ae68", "SIP_CODEC=g729") in new stack
   -- Executing Playback("SIP/213.5.1.6-40a2ae68", "the-party-you-are-calling|noanswer") in new stack
   -- Playing 'the-party-you-are-calling' (language 'gr')


However, the same dialplan without the noanswer option in the Playback, works fine. Here is the output that shows that g729 is used.

   -- Executing Set("SIP/213.5.1.6-40a2ae68", "SIP_CODEC=g729") in new stack
   -- Executing Playback("SIP/213.5.1.6-40a2ae68", "the-party-you-are-calling") in new stack
Feb 23 15:47:00 NOTICE[8095]: chan_sip.c:2506 sip_answer: Changing codec to 'g729' for this call because of ${SIP_CODEC) variable
   -- Playing 'the-party-you-are-calling' (language 'gr')
   -- Executing Playback("SIP/213.5.1.6-40a2ae68", "is-curntly-unavail") in new stack
   -- Playing 'is-curntly-unavail' (language 'gr')






****** ADDITIONAL INFORMATION ******

I am using the Asterisk-1.2.1 and I have applied the patch rtp-packetization-v2.diff.txt from http://bugs.digium.com/view.php?id=5162

Trying to solve the issue I created my own patch. I believe it solves the problem. However I am not 100% sure if there are any issues.

Here is the patch:

--- chan_sip.c  2006-02-13 14:58:52.768542984 +0200
+++ chan_sip.c.geop     2006-02-23 15:34:55.723163288 +0200
@@ -4521,6 +4521,22 @@
       respprep(&resp, p, msg, req);
       if (p->rtp) {
               ast_rtp_offered_from_local(p->rtp, 0);
+
+//////////////////////////////////////////////////////////////////
+               char *codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
+               if (codec) {
+                       int fmt=ast_getformatbyname(codec);
+                       if (fmt) {
+                               ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
+                               if (p->jointcapability & fmt) {
+                                       p->jointcapability &= fmt;
+                                       p->capability &= fmt;
+                               } else
+                                       ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+                       } else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
+               }
+/////////////////////////////////////////////////////////////////
+
               add_sdp(&resp, p);
       } else {
               ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);

Comments:By: Olle Johansson (oej) 2006-03-02 11:44:34.000-0600

You are right, we are not supporting SIP_CODEC for early media. I need a disclaimer for your patch proposal, it would be nice to have that on file - even not required since you're simply copying existing code to another place... :-)
By: George Papadopoulos (gpapadop73) 2006-03-03 01:38:20.000-0600

Do you mean it would be nice to have the patch on a file? I will upload it. Again, I am new to the "patching" business so I don't know if it is the right format. I created the patch in the channels directory, should I do it one directory up?
As for the disclaimer, I am not sure what you need from me. As you said, it is obvious that I just copied existing code.
By: Olle Johansson (oej) 2006-03-05 04:10:53.000-0600

Please try this patch that doesn't duplicate the code. Thanks for contributing and finding this.
By: Olle Johansson (oej) 2006-03-05 04:21:48.000-0600

Uploaded separate patch for svn trunk.
By: George Papadopoulos (gpapadop73) 2006-03-09 05:49:03.000-0600

I tried your patch and it works fine. Thank you.
Is this going to be in the next release, or will I have to apply the patch again?
By: Olle Johansson (oej) 2006-03-09 11:10:28.000-0600

Committed to Asterisk 1.2, revision 12477 and svn trunk (with a small modification for const char).
By: Olle Johansson (oej) 2006-03-09 11:10:56.000-0600

Thanks for a good and detailed error report, plus your testing!
By: Digium Subversion (svnbot) 2008-01-15 17:12:31.000-0600

Repository: asterisk
Revision: 11845

U   team/oej/test-this-branch/README.test-this-branch
U   team/oej/test-this-branch/channels/chan_sip.c

------------------------------------------------------------------------
r11845 | oej | 2008-01-15 17:12:31 -0600 (Tue, 15 Jan 2008) | 2 lines

Issue ASTERISK-6401 - Support SIP_CODEC for early media too

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http://svn.digium.com/view/asterisk?view=rev&revision=11845
By: Digium Subversion (svnbot) 2008-01-15 17:21:46.000-0600

Repository: asterisk
Revision: 12477

U   branches/1.2/channels/chan_sip.c

------------------------------------------------------------------------
r12477 | oej | 2008-01-15 17:21:46 -0600 (Tue, 15 Jan 2008) | 2 lines

Issue ASTERISK-6401 - SIP_CODEC not used for early media (reported by gpapadop73)

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http://svn.digium.com/view/asterisk?view=rev&revision=12477